This application claims priority to U.S. Provisional Patent Application Ser. No. 60/513,814, Filing Date Oct. 23, 2003, which is herein incorporated by reference.
1. Field of the Invention
This invention relates generally to a communications systems and methods. More particularly, this invention relates to systems and methods for bridging between types of communications networks. Even more particularly, this invention relates to systems and methods for bridging between a public switched telephone network and a packet switched digital communications network such as the Internet.
2. Description of Related Art
Referring to FIG. 1, the Public Switched Telephone Network (PSTN) 5, as is known in the art, provides an extremely reliable system that has been available for more than a century. It is simple to operate and is used at home and office where telephone sets 10 are connected directly to the PSTN network for communication. It offers a universally understandable telephone numbering plan i.e. Country Code, City or Area Code and the phone number. The standard PSTN offers several useful calling features such as Caller Identification Messages, Voice Mail and conferencing. The PSTN system, however, is not cost effective and does not offer the kind of features that are available when using a digital communication network such as the Internet.
The wireless telephone system 15 is not considered as reliable, but rather it offers mobility to the user through a cellular telephone 20. Alternately, the wireless telephone system maybe a satellite telephone system where the telephone 20 is not a cellular telephone but a satellite telephone. As structured today, the wireless system 15 is an integrated part of the PSTN 5 and consequently suffers from the same symptoms, i.e. lack of cost-effectiveness and accessibility features as those of the wired telephone system.
The digital communication network (Internet) 25 is a network of computer servers 30a, . . . , 30d that are interconnected with networking hardware and software. A computer 35 is connected through a single telephone connection commonly referred to as dial-up service or Digital Subscriber Lines (DSL) from the PSTN network 5 through a gateway 50 to a computer server 30b. An alternate connection service for a computer 35 to a computer server 30b is through a Synchronous Optical Network (SONET) network as marketed by cable television providers. Other network connections may be through networks such as a local area network such Ethernet, Fiber Distributed Data Interface (FDDI), Cable Modems, etc.
Computer telephony is a multi-media computer 35 running an Internet telephony software solution commonly referred to as Voice Over IP (VoIP) that takes advantage of the personal computer 30 as the hardware platform. NetMeeting (developed by Microsoft, Inc., Redmond, Wash.) and similar software packages are examples of computer telephony. The conversation typically takes place by using discrete speakers and microphone (not shown). Calling from one computer 35 to the other requires knowing static Internet protocol (IP) address of the other party. Otherwise, the parties need to register their dynamic IP address with a global computer server 30b. 
In an alternate configuration, the personal computer 35 is still required but an external device is a handset 40 connected via a network connection such as the universal serial bus (USB). Another configuration employs specially designed plug-in boards that support regular telephones are added to the computer 35. Some of these solutions continue to use software packages such as NetMeeting as their main engine. Even though this is a more attractive solution compared to the software only solution but it still suffers from the fact that the user requires a computer 35 to place a telephone call.
Another Internet appliance includes an embedded computer within a system and employs a telephone set. The embedded computer communicates with an Internet Service Provider (ISP) through the server 30b to communicate with a global computer server 30a, . . . , 30d executing and managing VoIP protocols. The attached telephone set functions essentially as the telephone 10 of the PSTN network 5, but communicates only with the global computer server 30a, . . . , 30d that manage and execute the VoIP protocols.
A similar appliance is referred to as a dial-up appliance. In this is a solution the embedded computer uses a modem and an ISP account to connect to the Internet 25 each time a telephone call is made. The main problem in this type of approach is the relatively long connection time, the inability to receive calls (the unit is not online at all times) and the undesirable extra modem delay.
Another development of the Internet appliance is the broadband appliance. In this instance, the Internet connection is broadband and thus the problems with the dial-up solution are resolved. The main issue here is the addressing scheme. Since there is no connection between the regular telephone number and the new appliance, a new number needs to be issued for incoming calls. The other problem stems from the fact that VoIP capability is only available for the telephone set, but not the landline. In fact there is no connection between the PSTN 5 connection and the broadband appliance.
The future of telephone appears at the present time to be Internet protocol telephones 45. The Internet protocol telephone 45 has an embedded computer similar to the above described appliances. The Internet protocol telephone 45 includes a built in handset and feature buttons. The embedded computer is structured to contain all the required hardware and software to appear to the user as a regular telephone, with the exception that the Internet protocol telephone 45 communicates through a computer server 30d to a global computer server 30a, . . . , 30d managing and executing the VoIP protocols. The Internet protocol 45 and the regular telephone 10 are totally independent, suggesting that the user should own both telephone sets to take advantage of both networks 5 and 25.
Gateways 50 are industrial networking equipment that bridge between the computer servers 30a, . . . , 30d and the PSTN 5 to provide access between the Internet 25 and the PSTN 5. For those computer servers 30a, . . . , 30d that provide the management and execution of the VoIP protocols, the Gateways are capable of handling many simultaneous VoIP calls. The Gateways 50 allow users to take advantage of the Internet telephony while being totally transparent to the user. This structure allows the VoIP communications to appear to the user as another long distance carrier.
Prepaid calling cards provide a user access to discounted providers of long-distance or toll calls services from home telephones 10 or public telephones by use of a toll-free number and password. The prepaid calling card reduces the long distance cost at home or when traveling. This method may or may not take advantage of VoIP. The main benefit of using calling card is to free the user from paying for long distance when the long distance tolls are high (making long-distance calls via cell phones, from hotels, airports, etc.).
Callback Systems are a means to pass a telephone number to a callback center to be called back on that number. The callback system is a popular device where the user commands a callback center to call a particular phone. Once the addressed phone starts ringing and the user picks up the phone, a special dial tone is heard and dialing can get started. Callback systems are useful when the user prefers to avoid any toll cost between the originating telephone and the long distance service provider. In other words the cost of the incoming call is less than the cost of the outgoing call.
Certain cellular telephone and personal digital assistant manufacturers are providing the ability for a wireless telephone 20 to communicate wirelessly with a computer server 30d as well as the wireless telephone network 15. The wireless telephone 20, operating in the wireless mode (WiFi) to communicate with the computer server 30d, uses a VoIP protocol for placing telephone calls through the digital communications network (Internet) 25. If there is no wireless connection, the wireless telephone 20 employs the wireless network 15 for placing telephone calls.
In summary, all of the above mentioned solutions can be categorized into three different groups:                PSTN only Solutions, such as regular telephones, wireless phones, prepaid cards, callback systems, etc. These solutions do not take advantage of the features provided by digital networks such as the Internet.        VoIP only solutions, such as IP phones, P.C.-based software packages, etc. These solutions assign virtual PSTN numbers in order to route the incoming calls. Therefore, the attributes of PSTN calls to real numbers (not virtual) are not transported to the Internet and vice versa.        Telephony Gateways. These devices function based on knowing the static IP address of the source and destination and subsequently no PSTN to IP address translation takes place.        It should be emphasized that none of the above solutions instantiates a PSTN-based telephone with other devices on the Internet. This inability originates from the fact that the existing IP solutions do not translate the PSTN attributes to the digital network and vice versa.        
U.S. Pat. Nos. 6,141,341 and 6,404,764 (Jones, et al.) describe an Internet Protocol telephone system and method that uses a telephone to place and receive voice over Internet Protocol (VoIP)-based telephone calls and public switched telephone network-based telephone calls. An off-hook condition with the telephone is detected and a sequence of signals generated by the telephone is received. At least a first signal generated by the telephone is buffered while the system attempts to detect a predetermined signal that signifies a VoIP-based call. Upon detection of the predetermined signal, the system intercepts subsequent signals in the sequence, absent the at least first signal that was buffered, and places the VoIP-based call via an Internet. Otherwise, the system places the PSTN-based call via a PSTN.
U.S. Pat. No. 6,141,345 (Goeddel, et al.) teaches a signal processing resource allocation for Internet-based telephony. An access platform pools different signal processing resources such as different types of speech-coding algorithms. The access platform is coupled to the Internet, a local-exchange-carrier (LEC), and other communications facilities such as long-distance facilities. For each call through the access platform, one of the plurality of signal processing resources is allocated as a function of signal type either through signal detection or out-of-band signaling. For example, the access platform first determines if the call is an audio call or data call by detecting the type of signal. If the call is an audio call, the access platform switches in echo canceling resources. On the other hand, if the call is a data call, the access platform determines if a speech-coding algorithm is being used and, if necessary, switches in a compatible speech-coding resource.
U.S. Pat. No. 6,353,660 (Burger, et al.) describes a call screening method allows a subscriber to screen calls made to the subscriber from callers using the PSTN while the subscriber uses another communications medium. An enhanced services platform (ESP) receives a first call from a caller using a particular public telephone number for the particular subscriber. The ESP identifies the particular public telephone number for the particular subscriber. The ESP accesses a database storing a public telephone number and a private packet-based address for subscribers to retrieve a private packet-based address of the particular subscriber on the basis of the particular public telephone number. An introductory message is provided to the caller and prompts the caller to leave a message. The ESP accesses the particular subscriber based on the particular subscriber private packet-based address to establish an audio connection via the communication medium. The subscriber is notified of the first call. If the subscriber answers the call, a communication path is provided between the caller and the subscriber via the communication medium so that the subscriber may hear the caller leave the message but the caller does not hear or know that the particular subscriber is listening. The ESP connects the caller and the subscriber for two-way communication upon the authorization of the subscriber. In another embodiment, both the caller and the subscriber use a packet-based network. In another aspect of the invention, the ESP records the caller's voice in response to the prompt, and plays the recording to the subscriber if the subscriber answers the call. The ESP provides a method for anonymously connecting an accessing caller to a subscriber using a packet-based network.
U.S. Pat. No. 6,560,457 (Silver, et al.) describes an enhanced call delivery system for interoperability between circuit switched and packet switched networks. The system expedites the delivery of a call originating in a circuit-switched network to a mobile terminal camped on a packet-switched network. Information representing the location of the mobile terminal in the packet-switched network is provided to the circuit switched network. A call setup with the mobile terminal is initiated with reference to the previously received location information. Location-based services are also provided by the circuit-switched network with access to such mobile terminal location information.
U.S. Patent Application 2002/0118671 (Staples, et al.) Aug. 29, 2002 illustrates a communication system that extends office telephony and network data services to remote clients through the Internet. The system has a telephony server, a local area network, a server system, and a user communication device. The telephony server (e.g. a Private Branch Exchange) provides telephony services for a plurality of office lines. The local area network couples to the Internet. The telephony server and local area network may reside within an office environment. The server system couples to the telephony server and to the local area network. The user communication device establishes a first connection to the server system through the Internet. In response to the first connection, the server system automatically provides access for the user communication device to the telephony server. Also, the server system automatically invokes a call forwarding operation in response to the first connection, so that subsequent telephone calls, intended to reach the user's office line, are forwarded to the server system. When the server system receives a first telephone call, which has been redirected by the telephony server from the user's office line, the server system forwards the first telephone call to the user communication device through the first connection. The user communication device may also establish a secure data connection to the server system through the Internet. The secure data connection provides the remote user with access to the local area network in a manner which protects the data security of the local area network.
“A Flexible, Module-Based Soc-Approach for Low-Power VoIP-Applications” Fugger, et al., 15th Annual IEEE International ASIC/SOC Conference, 2002, pp.: 256-260, September 2002 describes a system on chip (SoC) that is suitable for both conventional telephony via the public switched telephone network (PSTN) and voice over Internet protocol (VoIP). A highly flexible approach offers the possibility to integrate several interface types, customizable by software download. Additional download capabilities facilitate the integration of voice-codecs for voice compression, algorithms for silent noise injection, far end line echo cancellation, and fax/modem termination.
“Supplementary Services in the H.323 IP Telephony Network”, Korpi, et al., IEEE Communications Magazine, Vol.: 37, July 1999, pp.: 118-125 provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and inter-working of these services across hybrid networks.
U.S. Pat. No. 6,411,704 (Pelletier, et al.) describes a system and method for providing telephony services, such as caller ID, voice mail, selective call forwarding, etc. to a remote subscriber, typically located outside the provider's regional territory. The system can include customer premise equipment (CPE) selectively connectable to a remote central office (CO), a packet-switched network gateway, such as an Internet telephony gateway (ITG), for remotely communicating service requests to a local service node. The local service node can be located within the service provider's territory and returns telephony service responses via the packet-switched network gateway to the remote CO, and ultimately the remote CPE.
U.S. Pat. No. 6,377,570 (Vaziri, et al.) illustrates an Internet switch box that connects between a telephone set and a public switched telephone network (PSTN) line, the latter of which is used both for PSTN telephone conversations and for connection to an Internet service provider (ISP). The switch box contains hardware and embedded software for establishing a connection to an ISP and for Internet telephone. When two users, each having an Internet switch box connected to the telephone set, wish to have an Internet telephone conversation, one calls the other over the PSTN. When they agree to an Internet telephone conversation, they signal their Internet switch boxes, by pressing either buttons on the switch boxes or certain keys on the telephone keypads, to switch to Internet telephone. The switch boxes disconnect the PSTN call and connect to their ISPs. Once the switch boxes are on the Internet, they contact each other through a server which supplies Internet protocol (IP) addresses of switch boxes, and the users continue their conversation by Internet telephone. The users can also prearrange to call each other solely by Internet telephone, in which case they do not need to talk to each other over the PSTN.
U.S. Pat. No. 6,347,085 (Kelly) provides a method and apparatus for enabling communication between packet-switched data networks and circuit-switched communication networks utilizes the existing domain name system infrastructure of the Internet to resolve traditional PSTN telephone numbers into domain names, and, using one or more domain name servers, locate the network protocol address of a gateway capable of connecting an executing task on the packet-switched data network to the desired terminating apparatus on the circuit switched communication network. Also disclosed is a gateway architecture capable of performing the cross network connections as well as domain name server architecture that stores the segments of a telephone number, such as country code, area code and exchange, in a hierarchical tree configuration.
U.S. Pat. No. 6,304,565 (Ramamurthy) teaches a method for completing long distance pots calls with IP telephony endpoints. When a calling party calls a called party on the public-switched telephone network (PSTN), a database common to both the PSTN and an Internet Protocol (IP) network, is accessed to determine whether the called telephone line is currently active on the IP network. That database maintains a record for each user who is currently active on the IP network through an Internet Service Provider (ISP). In addition, each record contains information identifying the particular user logged into the network on that line, an indication whether the telephone line is IP telephony capable, an indication whether that line is currently active on an IP telephony call on an Internet Telephony network (ITN), and the IP telephony feature set subscribed to by the called party on that line. If the telephone line is active, IP telephony capable, currently active on an IP telephony call on the ITN, and the called party subscribes to a call waiting-on-IP feature, then the incoming call can be forwarded as an IP telephony call to the called telephone line over the ITN. Alternatively, the incoming call can be directed to an alternate destination on either the ITN network or the PSTN, such as a voice mail or message service. If the called party is engaged in a browsing activity and has an IP telephony capability, then the called party is alerted to the incoming telephone call via a message on his terminal.) The incoming call is then completed through the ITN to the called party as an Internet telephony call. If the called party is currently active on the IP network through an ISP, but does not have an IP telephony capability, then the PSTN forwards a message to the ISP, which pushes a message to the called party to inform him or her of the incoming call. The incoming call is then directed onto the IP network to a messaging service, or on the PSTN to a voicemail service, or to an alternate telephone number, such as a cellular phone, for pickup by the called party.
U.S. Pat. No. 6,278,707 (MacMillan, et al.) teaches a communication system that includes an interface between two networks, for example the public telephone system and an IP-based network. The system includes a modem bank which receives bearer channel inputs and provides outputs to the second network. A protocol converter interfaces with a signaling network such as a common channel signaling (CCS) network (e.g., an Signaling System 7 network). The protocol converter communicates signaling information to the modem bank. A resource manager is coupled to the protocol converter and includes a memory which stores status information relating to the system.
U.S. Pat. No. 6,134,235 (Goldman, et al.) illustrates a system and method for bridging the POTS network and a packet network, such as the Internet. The system and method use a set of access objects that provide the interfacing and functionality for exchanging address and payload information with the packet network, and for exchanging payload information with the payload subnetwork and signaling information with the signaling subnetwork of the POTS network. The system includes a communications management object that coordinates the transfer of information between the POTS network and the packet network; a payload object that transfers payload information between the system and the payload subnetwork of the first communications network; a signaling object that transfers signaling information between the system and the signaling subnetwork of the first communications network in accordance with a signaling protocol associated with the signaling subnetwork; and a packet object that transfers payload and address information between the system and the second communications network in accordance with a communications protocol associated with the second communications network.
U.S. Pat. No. 6,744,759 (Sidhu, et al.) details a system and method for providing user-configured telephone service to a user of a data network telephone. The user connects a data network telephone to the data network. The data network telephone registers with a telephone connection server to have basic calling service. The user accesses a service provider server to enter feature selections. The service provider server may use a web page to query the user for feature selections. The service provider server uses the user's selections to update the user's account and to activate the selected features.
U.S. Pat. No. 6,741,695 (McConnell, et al.) describes an interface engine that is communicates with a packet-switched network and to a legacy circuit-switched network. The interface engine uses a call processing protocol, such as the session initiation protocol (SIP), for packet-switched network communications and uses a legacy signaling protocol, such as IS-41, for legacy circuit-switched communications. In response to a registration request initiated by a subscriber device on the packet-switched network, the interface engine obtains a service profile for the subscriber from a call processing system, such as a home location register (HLR), of the legacy circuit-switched network. Service parameters derived from the service profile and stored in a service database may be used to apply services on the packet-switched network for the subscriber device.
U.S. Pat. No. 6,735,621 (Yoakum, et al.) provides a method and apparatus for messaging between disparate networks. A service control gateway (SCG) provides the capability to extend advanced intelligent network (AIN) services transparently between circuit switched and packet networks. Signaling system 7 (SS7) transaction capabilities application part (TCAP) messages are translated into session initiation protocol (SIP) INVITE messages. SIP messages, which may be responses to the translated messages referred to above, are translated back into TCAP messages. Data from messages is stored in an interaction database, a data structure maintained at the SCG. The SCG uses the interaction database to properly format translated messages for each network.
U.S. Pat. No. 6,714,536 (Dowling) teaches methods and apparatus for allowing a packet data connection to be established by sending an indication of a network address through a telephony path. In a first embodiment, a protocol stack initiates the establishment of an Internet connection by sending a data segment through a public switched telephone network (PSTN) telephony path and then operates and maintains the Internet connection on separate packet connection. Dialing digits are used to indicate the address of a remote computer or wireless device via the telephony path. This invention also enables mixed PSTN/Internet multimedia telephone calls. In an exemplary embodiment, when a point-to-point telephone PSTN connection is established, a screen of information automatically appears at one or both ends of the connection via the Internet.
U.S. Pat. No. 6,665,293 (Thornton, et al.) teaches an application for a voice over IP (VoIP) telephony gateway. The application provides a telephony gateway intended for paired use at opposite ends of a data network connection, in conjunction with at each end of a private branch exchange (PBX) for automatically routing telephone calls, e.g., voice, data and facsimile between two peer PBXs over either a public switched telephone network (PSTN) or a data network. The routing is based on aspects such as cost considerations for handling each call and called directory numbers, monitoring quality of service (QoS) then provided through the data network and switching (“auto-switching”) such calls back and forth between the PSTN and the data network, as needed, in response to dynamic changes in the QoS such that the call is carried over a connection then providing a sufficient QoS.
U.S. Pat. No. 6,650,901 (Schuster, et al.) describes a system and method for providing user-configured telephone service in a data network telephony system. The system and method provides location information and other information about a calling telephone to the caller during a telephone connection in a data network telephony system. Data network telephones may be provisioned and otherwise configured for operation with an extensive database and other user account information. The user's account may include a location identifier that identifies the physical location of the telephone. The location identifier may provide address information, latitude and longitude configuration, directions, or other information. The location information is communicated in a data communications channel between the caller and the called telephone. Alternately, the location information is communicated to the caller during call setup. The location information is useful in providing emergency dispatch services, such as 911 in a data network telephony system.
U.S. Pat. No. 6,628,617 (Karol, et al.) teaches a technique for Internetworking traffic on connectionless and connection-oriented networks. Traffic on a connectionless (CL) network, such as IP packets, can be routed onto a connection a connection oriented (CO) network, such as an ATM telephony network, when it is advantageous to do so from a user or service provider viewpoint, without affecting the ability of users to continue to use existing applications. Routing is controlled by nodes called CL-CO gateways, with connectivity to both the CL network and the Co network. When CL traffic originating at a source reached these gateway nodes, a decision is made whether to continue carrying the information in the CL mode, or to redirect the traffic to a CO network. Each CL-Co gateway may include hardware and software modules that typically comprise interfaces to the Co network, interfaces to the CL network, a moderately sized packet buffer for temporarily storing packets waiting for CO network setup or turnaround, a database for storing forwarding, flow control header translation and other information, and a processor containing logic for controlling the gateway packet handling operations.
U.S. Pat. No. 6,446,127 (Schuster, et al.) provides a system and method for providing user mobility services on a data network telephony system. User attributes may be transmitted from a portable information device, such as a personal digital assistant, to a voice communication device, such as an Ethernet-based telephone. The voice communication device receives the user attributes from the portable information device and may transmit a registration request to a registration server. The registration request may include the user attributes, and is used by the registration server to register the user to the voice communication device in a registration data base. When a call is placed to the user, the registration server may reference the registration data base to direct the call to the voice communication device.
U.S. Patent Application 2002/0107923 (Chu, et al.) illustrates a system architecture for linking packet-switched and circuit-switched clients. The system allows both phone-based and IP-based clients to participate in a single audio conference. At least two multi-point control units (MCUs) (i.e., conferencing servers) are enabled to connect via a standard data linkage (i.e., full-duplex dial-up or IP link). The system enables the phone-based MCU to handle the phone clients and the IP-based MCU to handle the IP-based clients, while connecting the two to allow each participating client to hear all other participating clients.
U.S. Patent Application 2004/0174864 (Klaghofer) describes a gateway device that is connected between a switched-circuit communication network and at least two VoIP network domains, each domain having a packet-based signaling controller. The gateway device transfers to each of the signaling controllers a registration request message so that the gateway device is simultaneously registered in the at least two VoIP network domains as a VoIP end point.
U.S. Patent Application 2004/0190498 (Kallio, et al.) teaches a method, system and gateway device that enables interworking between an IP-based network and a circuit-switched network. A first address information of a first connection end located in the circuit-switched network is routed in a trigger message from the IP-based network to a gateway control function. The first and second call legs are established in parallel towards the first connection end based on the first address information, and towards a second connection end located in the IP-based network based on a second address information obtained from the trigger message. A single connection between the two connection ends is then established by connecting the first and second call legs. Thereby, IP-based signaling functionality can be used to add capability for subscribers located in the CS domain to be invited into conferences or calls with subscribers located in an IP-based domain, e.g. the IMS domain.